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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_RTP_VIDEO_SENDER_INTERFACE_H_
#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
#include <map>
#include <vector>
#include "call/rtp_config.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
namespace webrtc {
class VideoBitrateAllocation;
struct FecProtectionParams;
class RtpVideoSenderInterface : public EncodedImageCallback {
public:
virtual void RegisterProcessThread(ProcessThread* module_process_thread) = 0;
virtual void DeRegisterProcessThread() = 0;
// RtpVideoSender will only route packets if being active, all
// packets will be dropped otherwise.
virtual void SetActive(bool active) = 0;
// Sets the sending status of the rtp modules and appropriately sets the
// RtpVideoSender to active if any rtp modules are active.
virtual void SetActiveModules(const std::vector<bool> active_modules) = 0;
virtual bool IsActive() = 0;
virtual void OnNetworkAvailability(bool network_available) = 0;
virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
virtual bool FecEnabled() const = 0;
virtual bool NackEnabled() const = 0;
virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
virtual void ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) = 0;
virtual void SetMaxRtpPacketSize(size_t max_rtp_packet_size) = 0;
virtual void OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) = 0;
};
} // namespace webrtc
#endif // CALL_RTP_VIDEO_SENDER_INTERFACE_H_
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